Protocolos para telefonia IP

Orientador: Nelson Luis Saldanha da fonsecaDissertação (mestrado) - Universidade Estadual de Campinas, Instituto de ComputaçãoResumo: A telefonia IP, também chamada de VoIP (Voice over IP), pode ser definida como qualquer aplicação telefônica usada em uma rede de comutação de pacotes de dados que utiliza o protocolo Internet Protocol (IP). Engloba novas aplicações que exploram a integração da comunicação de voz, imagens e de dados simultaneamente. Protocolos vêm sendo propostos para telefonia IP. No entanto, um grande desafio a ser transposto por estes protocolos é a garantia de qualidade de voz similar à da telefonia comutada por circuitos. Este trabalho apresenta os protocolos H.323, SIP, MGCP e Megaco/H.248 para telefonia IP, faz uma comparação destes protocolos e aborda fatores que afetam a Qualidade de Serviço (QoS) de telefonia IPAbstract: IP telephony can be defined as any telephonic application over the Internet Protocol and is one of the new applications that explore the integration of voice, image and data communication. Protocols have been proposed for IP telefony. However, one of the challenges in the IP telephony is to assure that the voice quality has similar quality of the one in circuit-switched telephony. This work presents the protocols H323, SIP, MGCP and MegacoIH.248 for IP telephony and compare them. It also describes the issues which impact the Quality of Service (QoS) in IP telephonyMestradoEngenharia de ComputaçãoMestre em Computaçã

Sinalização de media gateways em redes de próxima geração

Mestrado em Engenharia Electrónica e TelecomunicaçõesCom o grande crescimento das comunicações móveis e fixas, o acesso à Internet tornou-se cada vez mais numa exigência, colocando à industria das Telecomunicações, especialmente aos operadores, grandes desafios. Serviços comuns como chamadas de voz, podem agora ser oferecidos pelos Internet Service Providers (ISPs) aos seus clientes sobre a forma de serviço Voice over IP (VoIP). Este serviço deixou de ser exclusivo das redes Public Switched Telephone Network/Integrated Services Digital Network (PSTN/ISDN) e passou a ser fornecido também na Internet. Mas devido à necessidade de manter as tradicionais redes PSTN/ISDN, houve a necessidade de criar um ambiente de convergência, não só para estas redes mas também para outros tipos de redes de acesso, independentemente da tecnologia. É neste campo que os organismos de normalização e os operadores têm dado os seus contributos, criando uma rede de controlo e de transporte comum baseada em IP para a convergência de serviços. Inicialmente o 3rd Generation Partnership Project (3GPP) definiu uma arquitectura de convergência móvel com a rede IP, constituída por elementos de controlo, transporte e serviço, de nome IP Multimedia Subsystem (IMS). Mais tarde, esta arquitectura serviu de base (core) para o grupo TISPAN do European Telecommunications Standard Institute (ETSI) na normalização das Redes de Próxima Geração. Esta Dissertação pretende dar uma resposta à convergência fixo-móvel no âmbito da arquitectura PSTN/ISDN Emulation Subsystem (PES) do TISPAN. Este sistema permite que todos os clientes de uma Rede de Próxima Geração de um operador acedam a serviços das redes PSTN/ISDN e Digital Subscriber Line (DSL) de uma forma simples e imperceptível. Com este intuito foram desenvolvidos cenários de testes para os sistemas Trunking e de Acesso da arquitectura PES, tendo como objectivo final a sua integração na plataforma de próxima geração Service Handling on ip NETworks (SHipNET). Esta Dissertação experimenta várias situações reais de chamadas de voz sobre os cenários de testes, e inicia a implementação de um novo elemento definido para a arquitectura PES, Access Gateway Control Function (AGCF), para o controlo de Media Gateways nas redes de Acesso. ABSTRACT: With the big growth of mobile and fixed communications, Internet access has become a requirement, putting the telecommunication industry, and especially the operators, in front of a major challenge. Services such as voice calls can now be offered by Internet Service Providers (ISPs) to their customers. This service is no longer exclusive of Public Switched Telephone Network/Integrated Services Digital Network (PSTN/ISDN) and is now provided also through the Internet. But, because of the need to maintain the traditional PSTN/ISDN networks, there was a need to create a convergence, not only for these networks but also for other types of access networks, regardless of technology. The standards bodies and operators have made their contributions to create a network of control and transport policy, based on IP, for the services convergence. In the beginning the 3rd Generation Partnership Project (3GPP) defined an architecture for mobile convergence with IP network, made up of control, transport and service elements, called IP Multimedia Subsystem (IMS). Later, the core IMS served the ETSI TISPAN group in standardization of Next Generation Networks. This thesis aims to give an answer for fixed-mobile convergence within the architecture defined by TISPAN PSTN/ISDN Emulation Subsystem (PES). This system, formed by a Trunking, originally defined by the 3GPP IMS, and Access part, allows all customers of a Next Generation Network operator, access to PSTN/ISDN and Digital Subscriber Line (DSL) network services in a simple way. With this purpose, scenarios were developed for Trunking and Access systems of PES arquitecture, with the goal to integrate into the next generation platform Service Handling on ip NETworks (SHipNET). This thesis tests several real situations of voice calls on testing scenarios, and begins the implementation of a new element defined for PES arquitecture, Access Gateway Control Function (AGCF), for Media Gateways control purpose in access networks

Création de services de télécommunications en utilisant le protocole SIP et l'API JAIN-SIP

L'évolution du domaine des télécommunications et des réseaux d'Internet a abouti à l'intégration et la collaboration de ces deux domaines, ce qui a favorisé des résolutions de plusieurs problèmes de la téléphonie traditionnelle. Afin d'élargir les zones de services au-delà des frontières et des continents, les grandes compagnies de télécommunications et d'Internet ont investi dans le domaine des protocoles de signalisation dont H.323, MGCP (Media Gateway Control Protocol) et SIP. Depuis que l'Internet est devenu répandu, la téléphonie sur l'Internet (ou la téléphonie IP) est devenue populaire et fiable, ce qui a augmenté les demandes de services. SIP a été conçu pour remplacer le protocole H.323, étant plus simple et plus efficace. SIP est fondé sur le protocole HTTP. L'utilisation de SIP permet de concevoir des services de téléphonie IP qui s'intègrent facilement dans les réseaux informatiques. Comparativement à H.323 et MGCP, SIP est plus léger et plus simple. De plus, SIP est horizontal puisqu'il utilise des protocoles IP, ce qui le rend intéressant au niveau de l'implantation et de la création de services traditionnels et même des nouveaux services qui nécessitent de l'interaction avec les applications et les protocoles sur l'Internet. La motivation de ce travail vient de la croissance de la demande des services IP et le besoin de nouvelles méthodes plus simples et fiables s'intégrant aux réseaux de l'Internet afin de répondre à la résolution de la problématique de la création de services de télécommunications. Le premier objectif de ce travail est d'étudier et de proposer des solutions de conception de services dans l'environnement SIP. Le deuxième objectif est de proposer des solutions de réalisation et de mise en oeuvre de ces services

The possibilities of SS7 signalling transport over IP network using YATE switch

Diplomová Práce se věnuje problematikou signalizačního systému číslo 7 (SS7), a to zejména přenosem signalizace SS7 přes sítě na bázi IP protokolu. K ověření možností přenosu byla vytvořena konvergovaná síť s využitím Open Source PBX YATE, kde jsou potřebné protokoly implementovány. V úvodu diplomové práce je uveden popis signalizačního systému SS7, který je následován vysvětlením funkce každé z vrstev (MTP2 až aplikační) a přenášených zpráv síti SS7. Dále byla věnována pozornost protokolům, které umožňují přenos SS7 přes IP síť. V diplomové práci byla také popsána architektura PBX YATE, konfigurační soubory a způsoby instalace v operačním systému Linux. Rovněž byly stručně popsány důležité soubory pro realizaci této práce. Experimentální práce byly zahájeny s využitím dvojice virtuálních počítačů, které měly naistalovány dvě různé PBX, a to YATE a Asterisk. Dalším krokem byla realizace konvergované sítě pro ověření teoretických předpokladů. Pro tyto účely již byly využity servery s instalovanými TDM kartami. S pomocí těchto serverů byly ověřeny protokoly SS7, SIGTRAN, implementována MGCP brána a protokol SIP-T. Experimenty byly úspěšné, nicméně lze jistě pokračovat dalšími a ověřit další možnosti.This study examines the use of SS7 signaling system over IP networks by using the open source PBX YATE. At first it starts with describing the SS7 followed by an explanation of the function of each of its levels and the messages that are used within the SS7 network. The study then sheds some light on the ways of using SS7 inside IP network with the use of some protocols. It also discusses the architecture of YATE and its files, and how it is installed in Linux operating system. Finally, it describes the important files for delivering this task. The study was commenced by using two virtual machines that have two different open source PBX's which are YATE and Asterisk, and after acquiring some results by establishing communication between them via the means of SIP trunk, furthermore the study was extended to the laboratory in order to test it over real servers that have TDM cards, in order to apply the study by the means of SS7 protocols, SIGTRAN, MGCP gateway and SIP-T. The experiments have almost delivered successful communications after conducting a configuration for the files on multiple sides.

Theoretical study of molecular photoionization: diffraction and correlation effects

2013/2014Questa tesi raccoglie i risultati dell’attività di ricerca del mio dottorato che ha riguardato lo studio di molecole sottoposte a fotoionizzazione e il calcolo delle grandezze dinamiche coinvolte in questo tipo di processo. Una prima linea di ricerca ha seguito la descrizione degli effetti di interferenza e diffrazione nei profili di fotoionizzazione ad alte energie, attraverso un approccio basato sul metodo Density Functional Theory (DFT) accoppiato all’uso di una base di B-spline. Le oscillazioni derivanti da questi effetti di interferenza e diffrazione rappresentano un fenomeno universale, presente in tutte le molecole poliatomiche in esame, dalle biatomiche a quelle più complesse non simmetriche, dalla shell di core a quella di valenza più esterna. Nella regione di core abbiamo analizzato le oscillazioni presenti nel rapporto di intensità C2,3/C1,4 nello spettro di fotoelettone C 1s del 2-butino. Nella regione di valenza più interna abbiamo invece preso in esame gli spettri di fotoionizzazione di semplici molecole poliatomiche (propano, butano, isobutano e cis/trans-2-butene) e i risultati ottenuti sono stati confrontati con quelli sperimentali raccolti presso il sincrotrone Soleil di Parigi. Abbiamo poi analizzato l’effetto dovuto all’emissione coerente da centri equivalenti e quello dovuto alla diffrazione da atomi vicini non equivalenti negli spettri di core e di valenza. Nell’ambito di questa analisi, abbiamo preso in esame acetileni mono e disostituti con fluoro e iodio, comparando i risultati con quelli ottenuti nel caso del più semplice sistema acetilenico. Ci siamo inoltre occupati dello studio di effetti di intereferenza nella ionizzazione di valenza esterna di semplici idrocarburi e, nella stessa regione, abbiamo analizzato come la struttura geometrica di composti permetilati, in particolare la distanza metallo-anello, influenzi i loro profili di fotoionizzazione. Infine, nella regione di valenza interna, sono stati considerati i profili di ionizzazione per il caso di Ar@C60. I risultati sono stati messi a confronto con quelli ottenuti da uno studio precedente sulla molecola di C60. Una seconda linea di ricerca ha invece seguito la descrizione delle osservabili di fotoionizzaione considerando il contributo della correlazione elettronica. Questo può essere fatto attraverso l’implementazione di un formalismo closecoupling dove la funzione del continuo finale è espressa secondo un’espansione analoga a quella Configuration Interaction (CI) per gli stati legati. Il primo livello dell’implementazione ab initio è stato quello di descrivere accuratamente solo la correlazione negli stati legati. A questo scopo, sono stati utilizzati gli orbitali di Dyson. L’uso di questi orbitali è stato applicato alla descrizione delle osservabili di fotoionizzazione nel caso della molecola biatomica CS. Nello spettro di questa molecola è infatti presente un satellite ben risolto dovuto a effetti di correlazione elettronica che non possono essere descritti a livello DFT.The thesis is focused on the study of the dynamics of photoemission processes for atoms and molecules. A first line of research has followed the description of diffraction and interference effects in the photoionization profiles at high energy for several systems, through an approach based on the DFT method combined with the use of a B-spline basis. These diffraction and interference effects appear in the spectra as a result of wave propagation. The resulting oscillations represent a general phenomenon, present in polyatomic targets, from diatomics to complex non-symmetrical molecules, and from the deep core to the outer-valence shell. Firstly, in the core region, we analysed the oscillations in the intensity ratio C2,3/C1,4 in the carbon 1s photoelectron spectrum for 2-butyne. Then in the inner-valence shell region, the interference effects in the photoionization spectra of simple polyatomic molecules (propane, butane, isobutane and cis/trans-2-butene) were studied and the results have been compared with experimental data collected at the SOLEIL Synchrotron in Paris. Furthermore, we have analysed the effect due to coherent emission from equivalent centers and diffraction from neighbouring non-equivalent atoms in core and valence photoelectron spectra. For this, we investigated mono and disubstituted fluoro- and iodo-acetylenes and compared them to the simple acetylene system. We also focused on interference effects in the outer-valence ionization cross sections of simple hydrocarbons and, in the same shell, we also studied the influence of geometrical structures on photoionization profiles of permethylated compounds. Finally, in the inner-valence shell region, we considered the photoionization profiles for the case of Ar@C60. The results were compared with a previous study on the C60 molecule. A second line of research has followed the correlated description of photoionization observables. We developed a new method based on an ab initio closecoupling formalism. The use of the Dyson orbitals allowed to study the photoemission observables of highly correlated systems. As a first application of this method, we performed highly correlated calculations on the primary ionic states and the prominent satellite present in the outer-valence photoelectron spectrum of CS. Dyson orbitals are coupled to accurate one-particle continuum orbitals to provide a correlated description of energy-dependent cross sections, asymmetry parameters, branching ratios and Molecular Frame Photoelectron Angular Distributions (MFPADs).XXVII Ciclo198

Analysis and testing of voip-subsystems of Ip Brick

Technology in which communication done using IP(Internet Protocol) alternative for the traditional analog systems is voip (voice over internet protocol) .One of the emerging or attractive communication systems in this era is voip. Several technologies within the voip are emerging and more come to near future, services offered by this technology needs internet connections and/or telephone connections. It offers services such as making audio and video calls. VoIP is a medium that converts the analog signal to digital signals[1]. In this dissertation mainly focused on ipbrick voip-subsystems such as webrtc ,asterisk13.8 pbx server and kamilio sip proxy. These are free and open-source technologies available in the marketplace. Integration of these technologies provides real-time communication between the systems such as voice,video and text. Webrtc enables web browsers to have native support for real-time voice, video and data capabilities. As such, end-users do not require additional add-ons or plug-ins to utilize real-time voice and video communication. To allow webrtc to make calls to non-webrtc voip applications, A initiation protocol is needed and one such protocol is session initiation protocol (SIP),which is the standard protocol used for initializing, changing and terminating sessions for multimedia today. It is particularly known for its use in voip applications. Asterisk13.8pbx supports webrtc and it acts as media gateway. In this thesis, we are evaluating and comparing previous and current versions of the ipbrick voip-subsystem which is previously having lab environment of ipbrick OS v6.1 with Asterisk v1.8 as a pbx server and webrtc application such as webrtc2sip and SIP Proxy software called Kamailio which connects two endpoints. In this thesis we are testing ipbrick voip-subsystem with current version of ipbrick OS v6.2 with Asterisk v13.8 as a new pbx server and Webrtc application (SIPML5) on the browser side and a phone on the server side (softphone) establish a phone call between them. The SIP Proxy server Kamailio will act as a intermediary connection, connecting the two endpoints using websockets

Creation of value with open source software in the telecommunications field

Tese de doutoramento. Engenharia Electrotécnica e de Computadores. Faculdade de Engenharia. Universidade do Porto. 200

Design einer mobilen Anwendung zur verschlüsselten Sprachkommunikation auf Basis des Android Betriebssystems

Die unterschiedlichen Möglichkeiten und die Realisierung der abhörsicheren und verschlüsselten, mobilen Sprachkommunikation auf Basis des Android Betriebssystems, sind zentraler Bestandteil dieser Master-Thesis. Private und abhörsichere Kommunikation lässt sich in einem direkten Gespräch ohne größeren Aufwand realisieren. Bei indirekter Sprachkommunikation, wie einem Telefongespräch, muss erheblich mehr Aufwand betrieben werden, um abhörsicher und vertraulich kommunizieren zu können. Die Verwendung von Verschlüsselungsmechanismen ist hierfür eine Option zur Realisierung von abhörsicherer und privater Sprachkommunikation. In dieser Arbeit werden die verbreitetsten Protokolle und Technologien beschrieben und evaluiert, mit deren Hilfe man verschlüsselte prachkommunikation technisch realisieren kann. Die Technologie-Evaluation wird kategorisiert nach Netzzugangstechnologie, Audiocodec, Signalisierung, Medientransport und Schlüsselverwaltung durchgeführt. Dies geschieht unter Berücksichtigung von Angriffen und Sicherheits- lücken, den Besonderheiten der Medientransportebene bei drahtloser mobiler Datenübertragung und den Beschränkungen, denen mobile Endgeräte und der mobile Internetzugang unterworfen sind. Neben der Evaluation der Technologien wird auch der Einsatz und die Integration von Smartcards in Sprachkommunikationssoftware zur Verschlüsselung und Zertifikatsspeicherung diskutiert. Aufbauend auf der Analyse der existenten Protokolle und Technologien wird das Konzept einer Anwendung entwickelt, mit der die verschlüsselte Sprachkommunikation auf dem Android Betriebssystem realisiert werden kann. Dabei wird durch die Implementierung einzelner Teile des Konzepts, eine Machbarkeitsstudie durchgeführt.</p

Transmission média sur les réseaux IP en utilisant les protocoles SIP et IAX

Les progrès technologiques du réseau Internet ont permis le développement de nouvelles applications multimédia; la voix, la vidéo et la vidéoconférence sont devenues des domaines importants de recherche et de développement pour l’industrie des télécommunications. Ces dernières années ont été remarquables par la mise en oeuvre de connexion haute débit, et de terminaux mobile et fixe performants. Plusieurs standards ont été conçus spécifiquement pour permettre la transmission média sur les réseaux IP avec une meilleure qualité de service. Ce travail a pour but d’étudier les protocoles de transmission média sur les réseaux IP, en commençant par l’état de l’art de technologies principales pour accéder au réseau, les techniques utilisées pour encoder l’audio et la vidéo, et en finissant par les protocoles de transport combinés avec d’autres protocoles temps réels. L’objectif principal du mémoire est d’analyser, et intégrer les protocoles de transmission (SIP, RTP et IAX) sur les réseaux IP. Le projet se compose de deux parties : expérimentale et applicative. La première partie a pour objectif de mettre en place une plateforme IPPBX capable de fournir une solution assez complète de transmission média sur le réseau IP en utilisant les protocoles SIP et IAX. Ensuite, nous allons calculer le temps requis de signalisation SIP/IAX et la qualité de service d’une communication IAX en utilisant les codecs G.711 et GSM. La deuxième partie se compose de la conception et l’implémentation du protocole RTP dans les téléphones mobiles en utilisant la technologie J2ME pour permettre un environnement mobile de vidéoconférence. Nous allons effectuer un rapport technique assez complet décrivant la technologie mobile J2ME. Nous allons également tester les émulateurs et outils capables d’offrir un environnement de vidéoconférence mobile et les difficultés associées aux codecs supportés. Les résultats des expériences ont montré que le temps requis de signalisation SIP et IAX est sous un seuil acceptable dans un réseau local. Selon les valeurs obtenues du délai et de la gigue, la qualité de service de la communication IAX avec les codecs G.711 et GSM est adéquate. Le résultat obtenu de la partie applicative nous a permis de prouver que le client mobile de vidéoconférence est capable de s’enregistrer auprès d’un Proxy/Registrar pour joindre une session multimédia et de signaliser avec d’autres clients de la session via le protocole SIP. La conception du protocole RTP dans la technologie mobile adopte le RFC 3250 sur le plan théorique. L’architecture du système utilisé et les composantes logicielles ont été bien mises en place. La transmission des paquets RTP a été bien réalisée. La manipulation des paquets RTP en mode binaire a été bien effectuée pour rediriger les flux audio et vidéo au lecteur JMStudio

Estudio comparativo entre el software gestor Elastix 2.0 y 3CX 9.0 para la implementación de una central PBX Hosted

Estudio comparativo entre el software gestor Elastix 2.0 y 3cx 9.0 para la implemenación de una central PBX hosted en la Escuela de Ingeniería Electrónica en Telecomunicaciones y Redes de la ESPOCH. Se utilizó método deductivo, se trata del estudio y análisis de una tecnología que cumple con normas específicas y facilita recolectar información que permitirá dar solución al diseño de PBX. Se realizó análisis técnico para el estudio de prestaciones técnicas de Elastix 2.0 y de 3CX 9.0 éstas son: número máximo de usuarios, códecs soportados, costo de licenciamiento, funcionalidades de cada central, facilidad de adminstración y servicios adicionales. El análisis de licenciamiento por usuario establecido por cada grupo con relación 1-N obtuvo un ahorro del 100%. Elestix 2.0 es la opción más adecuada a nivel de servicios, prestaciones y costos de licenciamiento por presentar una efeciencia de 3 a 1 en relación a 3CX 9.0: Se desarrolló un documento guía para la implementación de Elastix 2.0 como gestor de un PBX hosted. Recomendándose que para la implementación de PBX hosted se siga la normativa planteada en este proyecto, ya que podrá ser desasrrollada en una empresa de telefonía IP o un IPS local con la infraestruictura adecuada